High-speed data communication paths are typically required in order to make high data rate services, such as video and Internet access, available to end users. Fiber optic communication paths are well suited for these high data rate services, but are not readily available in existing communication systems and are expensive to install. Therefore, data transfer technology which can operate with little modification to existing telephone wiring connection (e.g., twisted-pair copper lines) has been developed for providing high data rate services.
Since copper lines are widely available and developed, solutions to the high speed access problem were originally focused on improving the performance of voice-band modems, which operate at the subscriber premises end over a 3 kHz voice band and transmit signals through the public switching telephone network (PSTN). The phone company network treats them exactly like voice signals. These modems presently transmit up to 56 kb/s over 2-wire telephone line, even though the practical speed was 1.2 kb/s only twenty years ago. The improvement in voice band-modems over the past years has resulted from significant advances in algorithms, digital signal processing, and semiconductor technology. Because such modems are limited to voice bandwidth (3.0 kHz), the rate is bound by the Shannon limit, that is, the theoretical limit of error-free digital data that can be transmitted over a communication link with a specified bandwidth in the presence of noise interference—in this case 3 kHz. Thus, current voice modems have probably maximized the theoretical speed limits at about 56 Kb/s. However, there is a considerable amount of bandwidth available in copper lines that has gone unused by voice-band modems, and this is why a proposal known as Asymmetric Digital Subscriber Loop (ADSL) was suggested in the industry as a high-speed protocol/connection alternative over twisted-pair copper lines. The practical limits on data rate in conventional telephone line lengths (of 24 gauge twisted pair) vary from 1.544 Mb/s for an 18,000 foot connection, to 51.840 Mb/s for a 1,000 foot connection. Since a large proportion of current telephone subscribers fall within the 18,000 foot coverage range (approximately 3.5 miles), ADSL can make the current copper wire act like a much “bigger pipe” for sending computer bits and digital information (like movies and TV channels), while still carrying the voice traffic. For example, an ADSL modem can carry information 200 times faster than the typical 56K voice band modem used today.
ADSL is “asymmetric” in that more bandwidth is allocated to downstream data (to the subscriber) than upstream (back from the subscriber). The reason for this is a combination of cost, cross-talk, speed demands, and performance. For example, twisted pair wire coupling increases with the frequency of the signal. If symmetric signals in many pairs are used within a cable, the data rate and line lengths become significantly limited by the coupling noise. Since the preponderance of target applications for digital subscriber services is asymmetric, asymmetric bit rate is not perceived to be a serious limitation at this time. Therefore, the ADSL standard proposes up to 6 Mb/s for downstream, and up to 640 kb/s for upstream. For example, video on demand, home shopping, Internet access, remote LAN access, multimedia access, and specialized PC services all feature high data rate demands downstream, to the subscriber, but relatively low data rates demands upstream. The principal advantage is that all of the high speed data operations take place in a frequency band above the voice band, leaving Plain Old Telephone Service (POTS) service independent and undisturbed, even if an ADSL modem fails. ADSL further provides an economical solution for transmission of high bandwidth information over existing copper line infrastructures.
In order to adjust for performance variations over the entire available spectrum, ADSL systems typically use discrete multi-tone (DMT) techniques (e.g., a multi-carrier technique) that divide the available bandwidth of twisted-pair copper lines into multiple channels or “bins.” Using DMT, the available bandwidth is broken into a plurality of 4 kilohertz sub bands or tones and data is simultaneously transferred over several of the channels depending upon the transmission characteristics of each channel. For example, in conventional ADSL systems, the copper transmission line is divided into 256 DMT tones, separated by 4.3125 KHz, each with a different center frequency. Specifically, the T1E1.413 ADSL standard divides the available transmission bandwidth into two parts. At the lower 4 kHz band, ordinary (POTS) is provided. The bulk of the rest bandwidth in the range from 4 kHz to about 1 MHz is for data transmission in the downstream direction, which is defined to be from the exchange to the subscriber. The upstream control channel uses a 160 kHz band in between. The signals in each of these ranges can be extracted with an appropriate band-pass filter.
With DMT, a plurality of frames of a data stream is broken down into data blocks. Each data block is allocated to multiple carrier channels. A carrier channel, in turn, can be represented as a vector whose magnitude and phase is based on the data that the carrier channel is carrying and on the amount of bits that the carrier channel can support (sometimes referred to as “bit loading” or “tone ordering”). The bit loading of the carrier channel is indicative of the number of constellation points (e.g., the number of magnitude and phase combinations for the vector). Thus, if the bit loading of a particular carrier channel is 2, then the number of constellation points is 4, with a constellation point in each quadrant representing the binary number 00, 01, 10, or 11 for example. This process of associating binary numbers to constellation points is sometimes referred to as “constellation encoding” or “constellation mapping.” Each of the carrier channel vectors may be used to produce a quadrature amplitude modulated (QAM) signal at a given frequency. Each channel uses QAM to carry 2 to 15 bits/QAM symbol. The QAM symbols are then summed to produce a time domain DMT “symbol” that is subsequently transmitted over the twisted-pair copper line. That is, each of the carriers that make up the DMT symbol contains a QAM signal. A DMT symbol is generated for each frame of the original data stream. This results essentially in overall performance which is equivalent to around two hundred V.34 modems used in parallel on the same line. Because each carrier channel can be configured to a different bit rate according to the channel characteristics, it can be seen that DMT is inherently “rate-adaptive” and extremely flexible for interfacing with different subscriber equipment and line conditions.
In typical DMT implementations, such as shown in U.S. Pat. No. 5,479,447 to Chow et. al., hereby incorporated by reference in its entirety, transmission power to the individual channels is initially configured based on the noise power and transmission loss in each band. In this way, channels with less noise and attenuation can carry larger amounts of data, while poorer sub-channels can be configured to carry fewer bits and can even be shut down entirely. Information on the transmission characteristics of each sub-channel is typically stored in a bit and energy table as discussed, for example, in U.S. Pat. No. 5,596,604 to Cioffi et. al. hereby incorporated by reference in its entirety.
Initial line conditions may vary after initialization because of temperature fluctuations, interference, etc. This can affect both the error rate and maximum data throughput. Therefore, by measuring the quality of each sub-channel on an ongoing basis, an “updated” bit and energy table may be maintained to adaptively configure the system for maximum data throughput or error performance on an ongoing basis. In normal applications, if the quality of any particular channel degrades to the point where the error performance of the system is compromised, one or more bits on that sub-channel are automatically moved to a sub-channel that can support additional bits.
In order to improve the accuracy and performance of broadband modems, manufacturers began to design system to segment data depending on its characteristics and to transmit the information differently based on this segmenting—that is, some information may be less tolerant of errors than of latency, such as voice for example, while other information may require the least error prone transmission but be generally tolerant of latency, such as data for example.
In recognition of the differing accuracy and latency requirements of different data types transmitted over high speed data lines, dual latency techniques have been developed. In dual latency, multiple data paths are established over the available DMT channels—a fast path which has minimal latency but may contain errors and an interleaved path incorporating various forward error correction techniques to reduce the bit error rate (BER) for applications that require accuracy over speed.
As noted above, the interleaved data path often employs one or more forward error correction (FEC) techniques which, while adding latency to the data transmission, reduce the BER of the transmitted data. Typically, this involves addition of redundant information to the basic data, also known as payload data. The data bytes and redundant bytes together form a unit called a codeword. Redundant bytes are generally added on to the data bytes to form the last bytes of the codeword. The number of frames in a codeword is user selectable and is dependent on the number of payload bytes in a frame and the maximum size of a codeword. As a result of using a forward error correction techniques, a group of redundancy bytes are added to the S frames of payload data to form a codeword having a length of N bytes, which equals the number of redundancy bytes (R) plus the aggregate number of basic data bytes (K) for the S frames of data. The redundant bytes allows a degree of error detection and correction at the receiving end of the communication system.
Another forward error correction technique that may be employed in DSL systems in the interleaved data path is Reed-Solomon coding. Reed-Solomon coding works by first constructing a polynomial of the data symbols to be transmitted and then sending an over-sampled plot of the polynomial instead of the original symbols themselves. Because of the redundant information contained in the over-sampled data, it is possible to reconstruct the original polynomial and thus the data symbols even in the face of transmission errors, up to a certain degree of error. The advantage of using Reed-Solomon codes is that the probability of an error remaining in the decoded data is (usually) much lower than the probability of an error if Reed-Solomon is not used. This is often described as the coding gain. For example, a digital communication system is designed to operate at a BER of 10−9, that is no more than 1 in 109 bits are received in error. This can be achieved by boosting the transmission power or by adding Reed-Solomon or other FEC. Reed-Solomon allows the system to achieve this target BER with a lower transmitter output power. The power saving given by Reed-Solomon (in decibels) is the coding gain.
Yet another forward error correction technique employed in DSL systems is Trellis coding. The idea behind Trellis coding is that operations of modulation and coding are combined. The bandwidth is not expanded, that is, the symbol rate remains the same, but redundancy is introduced by using a constellation with more points than would be required without coding. The difference in signal-to-noise ratios between a coded and uncoded system of the same information rate that produced the same error probability is referred to as the coding gain. For a discussion of Trellis coded modulation refer to commonly assigned United States published patent application 2005/0010853 hereby incorporated by reference its entirety. Trellis coding may be used in both the fast and interleaved data paths.
A known problem with DMT-based DSL systems is noise. Noise in a DSL system can come in the form of additive white Gaussian noise (AWGN) attributable to outside sources and colored noise that can vary across the channel. Colored noise may be caused by cross-talk interference from adjacent signals in the twisted pair line. AWGN is white noise with a constant spectral density and a Gaussian distribution of amplitude. Before communicating over a DSL line, equipment at the central office (CO) as well as at the consumer premesis (CPE) will determined the current line conditions, that is the amount of noise on the line. Existing dual latency-based DSL systems estimate the received noise PSD at the provider's central office (CO) assuming the same noise margin for both fast and interleaved data paths. This results in a greater number of errors on the fast data path and an overall reduction in bandwidth.
Because AWGN and cross-talk noise can affect fast and interleaved data paths differently, existing solutions provide less than optimal performance. In practice, the signal-to-noise ratios for these paths will differ. Thus, there exists a need for improved systems and methods for resolving the signal-to-noise ratio (SNR) difference between fast and interleaved data path in dual latency xDSL systems.